The present invention relates to a data transmission system wherein an analog signal is digitized, the digital signal is applied to a transmitting system such as a transmission line, modulating/demodulating system, or recording/reproducing system, and the transmitted signal is received from the transmitting system and reconverted into the analog signal. More particularly, it relates to a signal transmission system which employs differential pulse code modulation (DPCM).
In recent years, owing to the advent of digital audio equipment etc., a transmission system has come into use wherein the analog signal of, e.g., music, is digitized by PCM (pulse code modulation), the digital signal is transmitted, and the transmitted signal is received and reconverted by a receiving system into the original analog signal. This system of digitizing the analog signal for transmission, however, has the problem that quantization noise develops during the PCM conversion of the analog singal. In order to reduce the quantization noise, the number of bits of the digital signal may be increased in the digitization of the PCM conversion. This measure, however, leads to the problem that the transmission efficiency is lowered.
On the other hand, in a case where the analog signal to be transmitted is an audio signal, there has been a system wherein conventional noise reduction circuits for analog signals are disposed before and behind the PCM transmission system because unless the noise is offensive to the ear, it can be, in effect, deemed to have been reduced.
FIG. 1 is a block diagram showing the system. In the drawing, the audio input, such as music applied to the input terminal Ain, is encoder in a noise reduction encodor 1. The encoded signal is digitized by an A/D (analog to digital converter) 2 and is subject to a data compression processing in a signal processing circuit 3 in a transmitting side.
The PCM signal transmitted from the transmission system is subject to a data expansion and data reproduction processing, in a processing circuit 4 in a receiving side. The data is again converted to the analog data in a D/A (Digital to Analog converter) 5 and is decoded in a noise reduction decoder 6 corresponding to the noise reduction encoder 1, thereby producing an audio output from an output terminal Ao. The audio output is produced, through a loudspeaker, as sound.
Quantization noise which develops during the digitization needs to be reduced for lowering the level of noise contained in the audio signal of music or other output to be emitted from a loudspeaker or other equipment. Regarding the reduction of the quantization noise, when the analog signal entering the A/D converter 2 exceeds the maximum value which can be processed by this converter, it is clipped to cause a clipping distortion. Therefore, the analog signal should preferably have a magnitude which is not greater than the maximum value processible by the A/D converter 2 and which is as close as possible to this maximum value. That is, the input signal of the A/D converter 2 should have a substantially constant value sufficiently close to the maximum value processible by the A/D converter 2. However, when the priorart noise reduction system is applied, a problem, which is discussed below, is involved. With the noise reduction system, the input signal should be rendered sufficiently greater than a fixed level of noise arising in the total system, such as a recording medium, and then transmitted (for recording, for example). It is ideal to perform this processing near the dynamic range of the transmitting system. On the side of the noise reduction encoder 1, accordingly, a great input signal is passed without change, and a small input signal has its level raised and is outputted. The latter signal is restored to the original magnitude on the side of the noise reduction decoder 6. In raising the level of the small signal in the encoding operation, however, this small signal must be controlled so as to hold a level lower than that of the great signal. Accordingly, even in the case where the encoded signals are applied to the A/D converter 2, the difference between the maximum and minimum values of the input levels of the A/D converter 2 cannot be made small. For this reason, the number of bits in the digitization cannot be rendered small in the prior-art noise reduction system. If the number of bits is set to be small, the influence of the quantization noise cannot be satisfactorily reduced.
In addition, the decoder side obtains control information in accordance with the magnitude of the level of the encoded signal, namely, the signal entering the decoder. In the case of the audio signal, in order to provide a proper signal level without incurring a ripple in a control signal even at the lower-limit value of the audio frequency, the control information needs to be obtained, as such low frequency is not followed up. To the end of obtaining the satisfactory control information from the input signal of the decoder, the period of time taken for obtaining the control information from the input signal, in other words, the control time constant, needs to be set at a sufficient value. The time constant of the encoder side is accordingly, also set at a large value. However, when the control time constant is increased in this manner, the control on the encoder side can not follow a high frequency signal, such as a signal of abrupt attack in the sound of a piano. In consequence, an excess level signal is applied to the A/D converter to give rise to a clipping distortion.
When the clipping distortion has arisen in the encoder side or the transmitting system due to the aforementioned cause, the relation of the magnitudes of levels can no longer be properly held in the transmission. As a result, not only is the clipped waveform reproduced on the decoder side, but even the unclipped part will have an improper level in many cases. This is attributed to the fact that correct control voltages cannot be obtained on the decoder side on account of the clipping phenomenon, and because of problems of tone quality, including the attack of a piano, take place.
In order to eliminate the drawbacks, a method to be described below has hitherto been utilized as a signal transmission system which can effectively reduce the noise component of the quantization noise that develops in the case of digitizing the analog signal.
Whereas the ordinary PCM encoding samples the original analog signal, such as an audio signal at every moment, and transmits the sampled value as digital data, namely, a PCM code, this method consists of a transmission system based on DPCM codes wherein only the difference between two successive sampled values is used as digital data. FIG. 2 shows an example of such transmission system employing the DPCM encoding.
In the illustrated system, the difference is not taken in terms of analog values but in terms of digital values. More specifically, on the sending side, the original analog signal is converted into digital data of, e. g., 15 bits, by an A/D (analog-to-digital) converter 7. The data enters a delay circuit 8 constructed of a register or the like, and is delayed by one sampling interval therein. The delayed signal enters a differentiator 9 along with the converted digital data of the original analog signal subsequently sampled. When the DPCM code consists of 16 bits, the differential data between both the digital data inputted to the subtracter or difference detector 9 is delivered to a transmitting system as data of 16 bits.
The differential data transmitted by the transmitting system enters a receiving side. On the receiving side, the differential data, which consists of 16 bits in this case, is applied to an adder 10. Here, a delay circuit 11 provides the output of the adder 10 corresponding to the last sampling interval, and the adder 10 adds up the differential data presently received and the last output of the adder 10 provided from the delay circuit 11. The added data which is composed of, e. g., 15 bits, is delivered to a D/A (digital-to-analog) converter 12, from which a reconverted analog signal is provided as an output.
The feature of the DPCM encoding is that, by transmitting the differential data between the respectively two samples adjoining in time, the values of the digital data to be transmitted can be made small on the average; in other words, the average level can be lowered, so that the bit number of the digital data can be reduced.
FIG. 3 is a diagram showing the relationship between the value D.sub.p of the data, in the case of transmitting the original analog signal S.sub.o by the usual PCM, and the value D.sub.d of the data in the case of trahsmitting the same by the DPCM. It is readily understood from the figure that, insofar as the sampling interval T.sub.s is of short (appropriate) signal period relative to the original analog signal, the transmission data has a smaller value on the average in the transmission by the the DPCM than in the transmission by the PCM. Especially in a case where the frequency of the original analog signal is sufficiently low in comparison with the sampling frequency as illustrated in the figure, the differential data D.sub.d becomes a much smaller value. Thus, if the quantization error is the same as in the PCM system, The DPCM system can bring a smaller number of bits into correspondence with the analog signal, and if the numbers of bits are equal, the DPCM system can reduce the quantization error. Therefore, the DPCM transmission becomes more effective.
Accordingly, when a certain music signal is transmitted, by way of example, as to the digital data to be transmitted by the DPCM transmission the appearance probability of smaller values is high and the appearance probability of larger values is conspicuously low, resulting in the rare appearance of large values, as shown in FIG. 4. In contrast, the tendency of the transmission data items to concentrate near zero is low in the PCM transmission. The feature of the DPCM transmission is based on the fact that, since the waveform of the music signal usually has a very gentle slope within a sampling time, and the difference between the two adjacent samples is comparatively slight, the digital levels concentrate near zero. Accordingly, when the PCM codes, each consisting of 16 bits, are transmitted by the DPCM, 8 to 10 bits or so are usually sufficient for properly transmitting most data items, and an appreciable transmission can be executed.
Although the transmission data value decreases on the average, such DPCM transmission has the problem that the maximum-level data appearing on rare occasions becomes substantially the same data value (level) as, or even more than that in the PCM transmission.
Thus, two characterizing features of the DPCM transmission are; (i) that the average level of the transmission data is very low, and, (ii) that although the maximum level of the transmission data is the same as, or more than in the usual PCM transmission, the probability of the appearance thereof is very low.
As a system for effectively transmitting the data items as above described, which has the low average level and which exhibits the low probability of the appearance of signals having great level differences, also to be considered is a system in which the ordinary transmission is executed with a predetermined number of bits smaller than that of the original data, whereas as to the signal having the great level difference in excess of the expressible range of the predetermined number of bits, only upper significant valid bits are transmitted with the predetermined number of bits, the remaining less significant bits being disregarded. In this case, as to the disregarded less significant bits, the contents of the disregarded bits are not sent, but only the number of the disregarded bits or the number of bits corresponding to the shift quantity is binary-coded and then transmitted to the receiving side. The aforementioned more significant valid bits are restored to the correct bit positions on the receiving side, whereby the original analog signal can be reproduced almost exactly. This system is called the "instantaneous companding pulse code modulation". In actuality, in each data block consisting of a plurality of samples, the maximum level value among the samples within the block is detected, the data within the block are shifted in accordance with the value and are incorporated into the data of the predetermined number of bits, and the resulting data is made the principal data. This system is called the "near-instantaneous companding pulse code modulation" (NIPCM). Along with the principal data, the binary data of the number of bits corresponding to the shift quantity, for example, is sent to the transmitting system as scale information once within one block interval. In this way, a substantially satisfactory analog signal transmission is permitted merely by transmitting one scale information for each data block which consists of the large number of sample data.
One example of such system is shown in FIG. 5 as a system block diagram of a transmission system.
In this case, an input analog signal comprising an audio signal is converted to an original data of a sufficient bit number, such as 15 bits, by a certain time interval, and then the maximum level or the level almost corresponding to the maximum level is detected, thereby providing 4 bit scale information. Then, the original data outputted from the A/D converter 13 is controlled in a digital level and is compressed to 8 bit data. Generally, the level control is conducted by shifting digits, and data compression is conducted by rounding off the lower significant bits. In case where the transmission data is 8 bits, and the scale information is 4 bits, a plurality of 8 bit transmission data, (for example, 32 samples) is composed, with four bit scale information in a composing means 16. The compressed data is transmitted to the transmission system.
When data is transmitted through the transmission line, one scale information is combined with a plurality of transmission data in a time-divisional manner.
The sampling frequency necessary for PCM and a frequency component included in the audio signal are considered. The sampling pulse SP shown in FIG. 6 has a very high frequency of 30 KHz.about.50 KHz in an ordinary case. A musical signal MS normally has a frequency component of 200 Hz to 3 Kz, and the level ME thereof changes at a conspicuously low frequency such as 0.5 Hz.about.300 Hz. Thus, even if a single scale information is combined with 32 transmission data, the amount of information of the scale information is not insufficient for transmission line, thereby enabling an efficient transmission. Even if, 100 transmission data are combined with a single scale data, data transmission can be conducted in a normal manner.
In the receiver side, 8 bit transmission data is separated from 4 bit scale data in separation means 17, and digital control of reverse processing namely, bit shifting from the transmission side is conducted, thereby producing an original data of 15 bits, which is converted to an analog data by the D/A converter 19, with the result of an output analog signal quite similar to the original analog data.
The above transmission system will be explained in more detail by reference to FIGS. 7A to 7C.
FIG. 7A shows the original data of 15 bits, in which the shaded portion is the effective bits.
In FIG. 7A, the effective data occupies 6 bits of the original data. Thus, the lower 8 bits of the original data can be made into the transmission data in an unaltered form. In this case, the digit of the eight bit of the transmission data is not changed from the lower bit side, and thus the scale information is "0" in this case. Thus, if the effective bit number is less than 8 bits, the scale information is uniformly kept "0".
In FIG. 7B, the effective data occupies 9 bits among the original data. In this case, if 8 bits are taken as transmission data as described above, the scale information is "1". The effective bit of the lowest digit, namely, LSB (the least significant bit) of the original data, is ignored. Such lower digits of the original data, which are ignored constitute error, namely, difference between the original data and the transmission data, but sufficiently small as compared with the transmission data.
In FIG. 7C, the effective bits occupy all of the original data, namely, 15 bits. In this case, the transmission data of 8 bits is located in such a position as to ignore the lower 7 bits of the original data. Thus, the scale information comes to "7". As is clear from this instance, the maximum amount of shifting is 7 bits.
Accordingly, the number of the scale information is 8 (2.sup.3) at a maximum and 3 bits is sufficient to express the scale data. The scale data can be compared with many original data of the prior stage with regard to many orignal data included in a predetermined period. The maximum value of many original data included in a predetermined period is measured or predicted in advance, thereby producing the common scale information (the amount of shifting) applicable to many original data. The scale information can be renewed at every plural data.
In the above description, the transmission data is composed only of data obtained by shifting bits of the original data. In case where analog signals to be processed have only one polarity, namely, either positive or negative, the transmission data may be an off-set binary code. In case where processed analog signals have both positive and negative polarities, a code bit or a bit corresponding to the code is included in the highest bit MSB of the original data (the most significant bit) and a 2's complement code is usually employed. This code bit is substantially an important bit and thus, the code bit of 1 bit should constitute the transmission data as the code bit of 1 bit is included in data obtained through the above-mentioned bit shifting. Thus, if the transmission data is 8 bits, one bit of the 8 bits should be the code bit in case of an audio signal.
In FIGS. 7B and 7C, the data to be added to the lower bits of the 8 bit transmission data in the receiving side is "0".
If the original data is expressed by 15 bits of 2's complement, namely, data as shown in FIG. 8A, the effective bits of 01100101 is extracted as the transmission data of 8 bits and the lower 4 bits are rounded off. Thus, 8 bits are transmitted and one scale information is transmitted at every 32 transmissions of 8 bit data. In the receiving side, the scale data which is transmitted once when the transmission data is transmitted 32 times, is separated at the separation means. The shifting of digits of 8 bit data within one block, namely, the varying of the digital level, is conducted based on the scale information. When the transmission data of 8 bits is subject to the bit shifting in accordance with the scale data, 0 data, (0, 0, 0, 0) is placed on the lower digits as shown in FIG. 8B, resulting in a large amount of error.
However, it is difficult that the conventional near instantaneous companding PCM transmission is applied to the DPCM transmission in an unaltered fashion, as the DPCM can achieve the data compression to a greater extent.
The reason is as follows: It is necessary to decode the received data by performing an integration of the received data. Thus, the error caused by rounding-off operation in the transmitting side is accumulated in the receiving side, thereby causing a great error to be produced. Thus, if the conventional near instantaneous companding system is applied to DPCM without modification, the actual transmission data includes a very large error, although it is intended to lower the average level and to decrease the accumulated errors.
Furthermore, in case a composite signal, formed of more than two signals, such as a higher frequency signal and a lower frequency signal, is inputted, the differentiate value of the higher frequency signal is larger, and the differentiate value of the lower frequency signal is small. The difference, namely, the difference between two differentiate values, are different in proportion to the frequency, even if two input signals have the same input level in amplitude. If the level of the lower frequency signal is smaller than that of the higher frequency signal, even by a small margin, the difference between the differentiate values increase. Thus, one scale information is determined by the high frequency signal and the effective bits of the lower frequency signal which should be transmitted fall into the bit position of digits to be rounded off, thereby sometime failing to be transmitted.